Let me start off with the environment that I am using. I have a virtual server that is hosted on Linode.com. They are running Xen Server VPS. The Freeswitch version I am using is FreeSWITCH Version 1.2.0-rc2+git~20120629T194106Z~5f09b40381+unclean~20120630T202857Z I am using mod_com_g729 with 20+ license installed. I have two Sip Profiles setup (internal and external). The internal with authentication is on port 5060 and external with no authentication on port 5080. They have different codec values for each. The Sip provider that I use is Flowroute.com The Leg from Flowroute to Linode I will call Leg A. The Leg from Linode to phones I will call Leg B. The problem that I have is that I can setup the pathway with the codec PCMU. I can place a call and both parties sound find and no audio problems. I can change the Leg A to g729 and still have no audio problems. I then change Leg B to g729 and the remote party (the one on the PSTN network) will hear the site traffic as choppy. The local party will hear quality issues not as much. It is a constant problem on the remote party side though. I then change Leg B back to PCMU and the quality is great. I change it back to G729 and it goes back to bad audio quality. All testing was done between 1am to 2am but this was also reproducible during the daytime. I'm pretty sure it is not a bandwidth problem. I have noticed a problem in the console that when one call is connected and stable this keeps scrolling over and over. http://pastebin.freeswitch.org/19466 Any ideas as to the cause of the audio problems with G729?